The present invention relates generally to the field of audio conferencing. More specifically, the present invention discloses a method for large-scale, fault-tolerant audio conferencing over a hybrid network.
The most common method to route calls for an audio conference is to control a local switch in a GSTN (globally switched telephony network). That is, a physical point-to-point connection is made between each piece of equipment in the network to create an overall point-to-point connection for the call. However, such a switch-controlled application can only route calls to devices connected to the switch, limiting the overall size of the system and limiting the geographic distribution of multipoint control units (MCUs) within the system. In addition, call transfer (e.g., from one MCU to another) requires that the connection from the switch to the new endpoint be established and the path to the transferring endpoint be torn down, thus limiting its use in a large-scale audio conferencing system.
Another conventional method to route calls for an audio conference is to interface with the network signaling layer (SS7/C7) directly, allowing for very large, geographically distributed systems. However, the difficulties of interfacing directly with the GSTN signaling layer prohibit all but the largest, most innovative audio conferencing system providers from implementing such a method.
Packet-switched call routing, on the other hand, facilitates dynamic call routing and call transfer during an audio conference. That is, no dedicated point-to-point connection is required in a packet-switched network. Each packet, including the call data and associated control, is sent individually to a destination address and the physical route taken from one endpoint to another can vary from packet to packet, eliminating the need for a dedicated circuit for each call. Thus, a call can be routed or even transferred within the packet-switched network simply by renegotiating the end point address. The ability to dynamically route and transfer calls between MCUs allows for greater geographic distribution of MCUs, permits an operator to service a large number of MCUs, and allows calls to be quickly switched between MCUs (e.g., to handle overflow) without interrupting service.
With existing circuit-switched networks (i.e., GSTN) and packet-switched networks becoming more commonplace, the need exists for audio conferencing over a hybrid network (i.e., a network that links both endpoints in a circuit-switched network and endpoints in a single conference system). In addition, a need exists to establish audio conferences in a conference system (i.e., to offer enhanced audio conferencing services) that is independent of the network that the endpoint is linked through.
There is a need for audio conferencing implemented over a hybrid network that provides both scalability and fault tolerance. Specifically, a need exists to monitor a pool of MCUs to determine which MCU can best handle the conference, and to dynamically route calls within the hybrid network so that a conference participant in one conference call can be transferred to another conference call and further, entire conferences can be transferred to other MCUs in the MCU pool without interrupting the audio conference (i.e., without tearing down connections and reestablishing the connections within the hybrid network). A need also exists for audio conferencing for receive-only or passive broadcast participants. Specifically, a need exists to provide a voice stream to the endpoints connected to the conference but that do not actively participate in the conference itself (i.e., do not contribute to the conference voice stream). Yet another need exists for full service audio conferencing using both high-touch (operator assisted) or reservation based audio conferencing and automated or xe2x80x9cad hocxe2x80x9d audio conferencing using the same platform. Specifically, a need exists to provide conferencing on a reservation basis and on an impromptu basis by monitoring a pool of MCUs to efficiently establish conferences over the hybrid network.
1. Solution to the Problem
None of the prior art references discussed above disclose large-scale, fault-tolerant audio conferencing implemented over a hybrid network.
This invention provides an audio conferencing method implemented over a hybrid network that provides scalability and fault tolerance.
A primary object of the present invention is to provide large-scale, fault tolerant audio conferencing using dynamically routed call transfer in a hybrid network. That is, the present invention monitors a pool of MCUs so that conferences can be efficiently established and routed to different MCUs when an MCU approaches capacity or when an MCU has to be taken out of service. As the audio conferencing method is implemented in a hybrid network, the destination of each audio packet can be rerouted seamlessly without interrupting the audio conference.
Another object of the present invention is to provide an audio conferencing method for receive-only or passive participants. That is, participants that do not actively contribute to the conference can be accommodated (i.e., receive the conference output or voice stream).
Yet another object of the present invention is to provide full service audio conferencing using both high-touch or reservation-based audio conferencing and automated or xe2x80x9cad hocxe2x80x9d audio conferencing on the same platform. That is, a conference need not be reserved against a dedicated MCU and instead, the method of the present invention allows a pool of MCUs to be monitored, thus allowing for both advance conference reservations and ad-hoc conferences.
2. Summary
The present invention discloses an audio conferencing method deployable in a hybrid network. Input is received from either or both circuit-switched endpoints and packet-based endpoints in a media gateway. The media gateway converts the input, if necessary, to an MCU-usable format and selects input based on predetermined selection criteria. An MCU mixes the selected inputs with other selected inputs to form an output stream and a sum stream matched with the endpoints in the audio conference. The output stream is a sum of the selected inputs from the plurality of endpoints exclusive of the input from the corresponding endpoint. The sum stream, on the other hand, includes the selected inputs. The media gateway converts the output stream and the sum stream to an endpoint compatible format, if necessary, and returns the converted output stream to the corresponding endpoint and the converted sum stream to the other endpoints connected to the audio conference. In one embodiment, both the media gateway and the MCU are part of a bridge server.